1. Field of the Invention
The present invention relates to a DTMF transfer method and a relay apparatus for transferring DTMF (Dual Tone Multi-Frequency) voice data on RTP (Transport Protocol for Real Time Applications) in an IP communication system using SIP (Session Initiation Protocol)(RFC3261 defined by IETF: The Internet Engineering Task Force).
2. Description of the Related Art
In recent years, VoIP (Voice over Internet Protocol) services are on the rise with the development of IP (Internet Protocol) networks. The VoIP services are techniques for transmitting/receiving voice data on IP networks. In VoIP, a virtual session is established between communication apparatuses. The IP-packetized voice data is transmitted over the established session. Session control protocols are required for controlling the establishment, maintenance, and disconnection of a session between communication apparatuses. Since SIP (Session Initiation Protocol) has high expandability of functions among them, SIP attracts attention as a session control protocol for VoIP. The SIP is an application protocol which uses a trans-interface mechanism, such as TCP (Transmission Control Protocol), UDP (User Datagram Protocol), etc. The SIP, which is a text-based protocols includes a header part for conveying a request or a response and a message body for describing the contents of a session. The description of an SIP session conforms to SDP (Session Description Protocol)(RFC2327 defined by IETF), etc.
In the following, a description will be given of the connection procedure by the SIP using FIGS. 1 and 2. FIG. 1 is an example of a system configuration for performing telephone communication in networks 20A and 20B through servers 30A and 30B using the SIP. The networks 20A and 20B individually include the SIP servers 30A and 30B, SIP terminals 41A-1 to 41A-3 and 41B-1 to 41B-3 connected to the SIP servers 30A and 30B through LANs 40A and 40B, and telephones 45A-1 to 45A-3 and 45B-1 to 45B-3 connected to the SIP terminals 41A-1 to 41A-3 and 41B-1 to 41B-3, respectively. The telephones 45A-1 to 45A-3 and 45B-1 to 45B-3 are connected under the SIP terminals 41A-1 to 41A-3 and 41B-1 to 41B-3. However, the apparatuses are not necessarily telephones. If the SIP terminal 41A-1 to 41A-3 and 41B-1 to 41B-3 can terminate telephone communication, for example, in the case where the SIP terminals 41A-1 to 41A-3 and 41B-1 to 41B-3 are SIP telephones, there may be nothing connected under the terminal.
FIG. 2 illustrates a sequence for establishing a session between the SIP terminal 41A-1 and SIP terminal 41A-2 in the same network 20. When the SIP terminal 41A-1 receives an [origination] signal from the telephone 45A-1 (S1), the SIP terminal 41A-1 transmits [INVITE], which is a message including the description of the addresses of the other party's SIP terminal 41A-2 to which a call is placed and the other party's telephone 45A-2 to the SIP server 30A (S2). The SIP server 30A transmits a temporary response, [100], to the SIP terminal 41A-1 which has transmitted [INVITE] (S3), identifies the location information of the SIP terminal 41A-2 from the description of the [INVITE], and transmits the [INVITE] for requesting call placing to the SIP terminal 41A-2 (S4).
The SIP terminal 41A-2 performs [calling] to the telephone 45A-2 (S5), transmits the temporary response, [100] (S6), and then transmits a temporary response, [180], to the SIP server 30A (S7). When the SIP server 30A receives the response, the SIP server 30A transmits a temporary response, [180], to the SIP terminal 41A-1 in the same manner (S8).
When the SIP terminal 41A-2 receives the [response] made by the telephone 45A-2 (S9), the SIP terminal 41A-2 transmits response [200], which is a message indicating the acceptance of a call to the SIP server 30A (S10). The SIP server 30A that has received the response [200] transmits the response [200] to the SIP terminal 41A-1 (S11). The SIP terminal 41A-1 that has received the response [200] transmits an acknowledgment [ACK], which is an acknowledgement message, to the SIP server 30A (S12). Similarly, the SIP server 30A transmits the acknowledgment [ACK] to the SIP terminal 41A-2 (S13).
As described above, a session between the SIP terminal 41A-1 and the SIP terminal 41A-2 is established (S14), and the telephone communication between the telephone 45A-1 and the telephones 45A-2 by RTP (Transport Protocol for Real Time Applications) (RFC1889/RFC3350 defined by IETF) becomes possible (S15). In general, the call placing request, [INVITE], and the response [200] include information (session information) for transferring a voice packet between the SIP terminal 41A-1 and the SIP terminal 41A-2. The SDP, etc., is used for the description of the session information. By conforming the specifications of the SIP and the SDP, it is possible to specify the SIP terminal information and the SIP server information by an IP address. Also, the session information is sometimes included in the temporary response [180], and it is possible to transmit/receive voice before the response [200].
On the other hand, since IP networks have become widespread rapidly, a technique for the interconnection between areas (networks) having different IP address (in the following, simply called “address”) systems and the interconnection of SIP protocols having original headers, etc., becomes necessary. For the interconnection between areas having different address systems, the problem has been able to be solved by using an apparatus for converting addresses as disclosed in Japanese Unexamined Patent Application Publication No. 2003-174466.
A description will be given of the case where an apparatus having a function of recognizing and transmitting dial (a push-button signal constructed by DTMF) by DTMF (Dual Tone Multi-Frequency) voice data such as a telephone, a PBX (Private Branch Exchange), etc., is connected under an SIP terminal in a system configuration using an address-conversion apparatus using FIGS. 1 and 3. Here, an SIP converter 10 in FIG. 1 is assumed to be an address-conversion apparatus between the networks having different address systems. FIG. 1 is an example of a communication system for performing telephone communication using SIP. FIG. 3 illustrates a sequence of dial transmission/receiving after a session is established between the SIP terminal 41A-1 in the first network 20A and the SIP terminal 41B-1 in the second network 20B.
In a system in which a session has been established by SIP, there are two types of methods for performing transmission/receiving dial (a push-button signal constructed by DTMF). The first method is a method in which the DTMF is packetized as voice data and transmitted by RTP similarly as usual voice data without the detection of DTMF by an apparatus terminating SIP. The second method is a method in which the DTMF detection is performed, the detected DTMF is converted into coded DTMF information (coded DTMF information different from voice data) in a predetermined format, and is transmitted to an opposite apparatus. The opposite apparatus that has received this coded DTMF information decodes this into DTMF as voice data to reproduce the data or recognizes the transmitted DTMF based on the coded DTMF information.
As an example of the second method, there is a method in which DTMF is transmitted using the [INFO] method in SIP. In this regard, the [INFO] method is defined as an RFC2976 by IETF. In the [INFO] method, information such as the type of (push-button information of 0 to 9, #, *, A to D) DTMF to be reproduced, DTMF reproduction time, etc., are described as coded DTMF information. When an apparatus terminating SIP (DTMF transmission apparatus) detects DTMF, the DTMF is converted into the [INFO] method, which describes the type of DTMF corresponding to the detected DTMF and the reproduction time reproduction time, and transmits it to the opposite apparatus. The opposite apparatus (DTMF receiving apparatus) converts the coded DTMF information (the type of DTMF and the reproduction time) in the [INFO] method into the DTMF as the original voice data.
Also, as another example of the second method, there is a method in which when DTMF is detected, the coded DTMF information corresponding to the detected DTMF is stored in the payload area of the RTP, and is transmitted to the opposite apparatus (DTMF receiving apparatus). This method is defined as RFC2833 by IETF, and includes a description of information such as the type of DTMF to be reproduced (push-button information of 0 to 9, #, *, A to D), the DTMF reproduction time, etc., as the coded DTMF information. The opposite apparatus (DTMF receiving apparatus) converts the coded DTMF information (the type of DTMF, the reproduction time, etc.) in RTP into the DTMF as the original voice data.
First, a description will be given of the first method using FIG. 3. While a session is established between the SIP terminal 41A-1 of the first network 20A and the SIP terminal 41B-1 of the second network 20B (S21), when the SIP terminal 41A-1 receives DTMF from the telephone 45A-1 (S22), the DTMF is not detected, and the voice data corresponding to the DTMF is transmitted to the SIP terminal 41B-1 as voice data on RTP in the same manner as normal voice (S23). The SIP terminal 41B-1 notifies the voice data (DTMF) in the RTP to the telephone 45B-1 (S24).
A description will be given of the second method using FIG. 3. In this regard, in the following description of the present invention, the term “coded DTMF information” is used by defining as coded information produced by coding (a form in which information such as the type of DTMF, the reproduction time, etc., is represented by data) DTMF as voice data into a predetermined format different from the voice data using the [INFO] method based on RFC2976 by IETF or “RTP storing coded DTMF information” based on RFC2833 by IETF. Accordingly, “coded DTMF information” is used as information different from the voice information produced by storing DTMF into RTP as voice data.
While a session is established between the SIP terminal 41A-1 of the first network 20A and the SIP terminal 41B-1 of the second network 20B (S21), when the SIP terminal 41A-1 detects DTMF from the telephone 45A-1 (S31), the coded DTMF information containing the description of the type of the detected DTMF and the reproduction time is created, and the coded DTMF information is transmitted to the SIP server 30A (S32). The SIP server 30A transmits the coded DTMF information to the SIP converter 10 which is an apparatus for performing address conversion (S33). The SIP converter 10 performs address conversion between the networks having different address system on the transmitted coded DTMF information (S34), and transmits the coded DTMF information containing the description of the type of the detected DTMF and the reproduction time of the second network 20B to the SIP server 30B (S35). The SIP server 30B transmits the coded DTMF information to the SIP terminal 41B-1 (S36). The SIP terminal 41B-1 determines the type of DTMF and the reproduction time from the coded DTMF information, and transmits the DTMF to the telephone 45B-1 using the DTMF reproduction apparatus including a DTMF-signal transmitter, etc. (S37).
When the second method is used, if the SIP terminal 41B-1 of the second network 20B does not have the DTMF reproduction function corresponding to the coded DTMF information (that is to say, if the SIP terminal 41B-1 does not support the [INFO] method processing based on RFC2976 by IETF or “RTP storing coded DTMF information” processing based on RFC2833 by IETF), the received coded DTMF information is not reproduced to DTMF. Thus, if an apparatus, which transmits and recognizes DTMF, such as a telephone, a PBX, etc., is connected under an SIP terminal, problems sometimes arise in that information is not normally transferred, a telephone is not connected, transfer is not possible, etc. In this manner, when the SIP terminals connected to an address conversion apparatus include an apparatus supporting the [INFO] method processing based on RFC2976 by IETF or “RTP storing coded DTMF information” processing based on RFC2833 by IETF and an apparatus not supporting the processing, there have been various problems so far.
Accordingly, it is an object of the present invention to provide a method for DTMF transfer which allows normal DTMF transmission between SIP terminals even if a communication system includes an SIP terminal which supports the [INFO] method processing based on RFC2976 by IETF or “RTP storing coded DTMF information” processing based on RFC2833 by IETF and an SIP terminal which does not support the [INFO] method processing based on RFC2976 by IETF or “RTP storing coded DTMF information” processing based on RFC2833 by IETF.